The present invention relates to hearing aids, and more particularly to hearing aid signal processing circuits having a feedback reduction algorithm.
The primary components of any body worn hearing aid include a microphone which converts an acoustic sound into an electrical signal, an amplifier which increases and often modifies the electrical signal, and a speaker (commonly called a receiver) which converts the amplifier output into a generated acoustic sound. Because the sound output from the receiver propagates three-dimensionally, a basic difficulty of hearing aids occurs when the generated acoustic sound from the receiver travels back as a sound wave received by the microphone. If the amount of amplification gain is sufficient and the internal and external delays result in a multiple of 360° at a given frequency, the feedback loop will become unstable, wherein the each loop of signal (electrically forward through the amplifier, and then with a portion fed acoustically back through physical space) increases. Acoustic feedback is well known as a loud and annoying whistle or buzz heard in public address systems, and is also well known as loud and annoying whistles, squeals chirps or buzzes generated in hearing aids.
To attempt to minimize acoustic feedback, one strategy is to fit the hearing aid as tightly as possible into the ear canal between the microphone and the receiver, and therefore physically minimize the amount of transmission of the acoustic signal from the receiver back to the microphone. However, tight fits are difficult to achieve and uncomfortable. Further, tight fits in the ear canal result in an “occlusion” effect, and many hearing aid wearers will complain that the occlusion effect prevents them from properly hearing their own voice in a natural way with the hearing aid.
With the difficulties in stopping the physical transmission of acoustic feedback, processing strategies of the electrical signal within the hearing aid are also used to reduce feedback. Accordingly, feedback cancellers are an important part of modern hearing aid design. By canceling the annoying squeal of feedback, hearing aids are more desirable to wear. Feedback cancellation processing algorithms allow more gain and a less tight fit that is a more comfortable fit in the ear canal.
An older feedback cancellation technique uses a notch filter in the forward path to reduce the gain at the offending frequency. Feedback typically occurs in the 1 to 7 kHz range. Unfortunately, this is also a frequency range carrying much sound information that most users desire to hear. This notch filter technique is now seldom used in hearing aids since it results in a reduction of gain in a frequency range where gain is desired to best improve the hearing results of the wearer.
More modern feedback cancellation is made possible with digital amplifiers and processors, and involves phase cancellation. The basic premise of phase cancellation is to attempt to determine which portion of the electrical signal being processed by the amplifier occurs from the external acoustic/physical feedback loop, and then to add a processor-generated portion, opposite in phase, and timed to the electrical signal so the added portion exactly and oppositely removes or cancels the transmitted acoustic feedback known to occur.
If it was possible in real time and at low cost to accurately identify which portion of the microphone signal resulted from feedback, it would be relatively easy to have the algorithm remove acoustic feedback. A first problem is that neither the incoming (non-feedback) acoustic signal nor the exact magnitude of the feedback acoustic signal is known a priori. Different acoustic conditions, such as changes in the physical fit of the hearing aid in the ear canal, sound reflective surfaces near the ear, changes in the physical shape of the ear canal during jaw movement, etc., will change the magnitude and frequency profile of the feedback acoustic signal. A second problem is that the exact time for the acoustic feedback signal to travel from the receiver to the microphone is not known. The amount of time required for the acoustic signal to directly travel the distance from the receiver to the microphone is reasonably short (on the order of 20-100 microseconds), and can be closely determined for the geometry of any particular hearing aid. A greater delay is typically introduced in the electrical processing time of the signal (typically 2-10 milliseconds), which can be precisely determined for the processor in any particular hearing aid. While sampling rates of 40 kHz or more are needed to fully reproduce the spectrum of human hearing, sampling rates for hearing aids are more commonly around 16 or 20 kHz, providing a bandwidth of 8 or 10 kHz. At a sampling rate of 16 kHz, a 6 millisecond processing time and a typical in-the-canal hearing aid geometry, the acoustic feedback from the output will be picked up about 100 samples later. However, feedback is significantly affected by sound reflection (off a telephone hand set, etc.), and the time delay for the echo wave depends upon the instantaneous location of the sound reflective surface, which is not known a priori and can change quickly. That is, the magnitude and frequency profile (a/k/a the feedback transfer function) and the time delay of sound transmitted in the acoustic feedback channel are not constant, but rather will change during events and conditions of the hearing aid.
Phase cancellation is achieved in practice with an internal filter (typically finite impulse response, or “FIR” filter) that is adaptively adjusted such as with a least mean squared controller to mimic the external acoustic feedback path. Subtracting the output of the internal filter from the input signal from the microphone results in significant cancellation of the acoustic feedback while maintaining the desired forward gain. For this design to work well, the internal FIR filter must match the external path in both amplitude and phase (exact feedback loop delay timing) for all frequencies that are potential problems.
For the vast majority of circumstances, modern phase cancellation algorithms can increase the stable gain of a hearing aid up to 15 or 20 dB beyond the stable gain without feedback cancellation. However, circumstances in which the input is “self-correlated” pose special problems. In a self-correlated signal, such as music, ringtones or other periodic sound inputs, the incoming acoustic signal in a particular frequency band may be identical both before and after the feedback loop delay, such that the incoming acoustic signal is completely indistinguishable from the acoustic feedback signal in the microphone output. In addition to having an adaptive FIR internal feedback filter, modern phase cancellation algorithms can include subroutines to modify the method of adjusting the FIR coefficients for occasions when the input is self-correlating. For example, U.S. Pat. No. 7,519,193 owned by the assignee of the present invention and incorporated herein by reference, discloses a phase cancellation algorithm wherein a correlation detector and a phase shifter are used in the feed forward path to attempt to determine which portion of the incoming signal is due to external acoustic feedback and to attempt to prevent phase matching between the external acoustic feedback signal and the incoming signal even when the incoming signal is self-correlating.
While feedback cancellation algorithms can greatly increase permissible gain, they still have trouble adaptively correcting during times when the external feedback path is rapidly changing, such as raising one's hand or a telephone receiver to the ear, combing one's hair over the ear, during jaw motion or other events. Even with advanced adaptive feedback cancellation algorithms, some feedback squeal can be heard when the external feedback path rapidly changes. This feedback squeal can result either from classical feedback, or as “entrainment” when the correction generated by the feedback cancellation algorithm (instead of acoustic feedback of the original signal) actually creates an audible artifact in the output. Attempts to identify and correct the entrainment problem are disclosed in U.S. Patent Pub. Nos. 2003/0026442 and 2005/0036632, incorporated by reference.